Telecommunications networks currently rely to a large extent upon the Signalling System no.7 (SS7) as the mechanism for controlling call connections and for handling the transfer of signalling information between signalling points of the networks. Typically, one or more application and user parts at a given signalling point will make use of SS7 to communicate with peer application and user parts at some other signalling point. Examples of user parts are ISUP (ISDN User Part) and TUP (Telephony User Part) whilst examples of application parts are INAP (Intelligent Network Application Part) and MAP (Mobile Application Part). The conventional SS7 protocol stack includes Message Transfer Parts MTP1, MTP2, and MTP3 which handle the formatting of signalling messages for transport over the physical layer as well as various routing functions.
There has been considerable interest of late amongst the telecommunications community in using non-standard (i.e. non-conventional within the telecommunications industry) signalling transport mechanisms in telecommunications networks in place of the conventional SS7 mechanisms. The reasons for this are related both to improvements in efficiency as well as potential cost savings. Much consideration has been given for example to the use of Internet Protocol (IP) networks to transport signalling information between signalling points. IP networks have the advantage that they make efficient use of transmission resources by using packet switching and are relatively low in cost due to the widespread use of the technology (as opposed to specialised telecommunication technology). There is also interest in using other transport mechanisms including AAL1/2/5, FR etc.
The ISUP standard which deals with the setting-up and control of call connections in a telecommunications network is closely linked to the SS7 signalling transport mechanism and does not readily lend itself to use with other non-standard transport technologies such as IP and AAL2. As such, several standardisation bodies including the ITU-T, ETSI, and ANSI, are currently considering the specification of a signalling protocol for the control of calls, which is independent of the underlying transport mechanism. This can be viewed as separating out from the protocol, Bearer Control functions which relate merely to establishing the parameters (including the start and end points) of the “pipe” via which user plane data is transported between nodes, and which are specific to the transport mechanism. The new protocol, referred to as Transport Independent Call Control (TICC), retains Call Control functions such as the services invoked for a call between given calling and called parties (e.g. call forwarding), and the overall routing of user plane data.
The new network architecture resulting from the separation of the call and Bearer Control levels results in an open interface appearing between a Call Control entity and a Bearer Control entity, where these entities are referred to as a Media Gateway Controller and a Media Gateway respectively. The open interface is referred to hereinafter as X-CP, examples of which are the MEGACO work of the IETF and the H.248 work of ITU Study Group 16 (SG16).
Traditionally, fixed telephone networks make use of Pulse Code Modulation to transport user plane data, e.g. voice, facsimile, etc, between network nodes. Modern cellular networks on the other hand often use one or more coders/decoders (referred to as “codecs”) to compress voice signals for efficient transmission across the air interface and within the cellular networks themselves. Where a telephone call connection extends between two networks (or terminals) which support different or multiple speech codecs, a negotiation may be carried out between the terminals to decide upon an appropriate codec. If this negotiation is not carried out, the result may be a requirement for transcoding at the interface between the networks, i.e. conversion from one form of speech coding to another. Transcoding is expensive in terms of resources, significantly degrades speech quality, and introduces a processing time delay. Codec negotiation is therefore the preferred option.
In addition to codec negotiation, there is often a need in conventional telecommunications networks to negotiate other functionality and parameters. For example, it may be desirable to negotiate security capabilities such as voice ciphering and data encryption between terminals or nodes in telecommunications networks.